This invention relates to digital speech compression systems, and in particular to systems of that type that utilize linear predictive coding techniques.
A great deal of current research in the area of narrow band digital speech compression makes use of some form of linear predictive coding (LPC) to extract on the order of 10 to 12 parameters approximately 50 times a second to specify the speech spectrum. In a typical implementation, the input speech waveform in analog low pass filtered to about 3200 Hz, analog-to-digital converted at about 6400 Hz with LPC analysis done in a digital signal processor using 16 bit fixed-point arithmetic.
The most time consuming data processor tasks are the formation of correlation coefficients (usually using double-precision accumulation) in the analyzer, and the synthesis of the output speech using a recursive filter. Although it has been recognized that reduction of the computation load for analysis and synthesis in these systems would provide improved fidelity and also permit the use of smaller and cheaper computer processing units, no effective means for such reduction have yet been devised. The present invention is directed toward achieving such an improvement in linear predictive coding systems.